Video telephony is thought to be a promising service offering, with many telecommunications and cable companies sponsoring trials. The video streams in video telephony communications are carried over internet protocol (IP) networks. In order to conserve bandwidth, the video data is compressed using efficient video coding standards, such as the International Telecommunication Union (ITU-T) H.264 standards (also referred to as MPEG-4 Part 10 or Advanced Video Coding (AVC)). H.264 exhibits a combination of new techniques and increased degrees of freedom compared to those used in existing compression algorithms, such as H.263, MPEG-2, and MPEG-4 (simple profile). Among the new techniques defined in H.264 are 4×4 pixel macroblocks, Integer Transform to replace the Discrete Cosine Transform, multi-frame prediction, context adaptive variable length coding (CAVLC), SI/SP frames, and context-adaptive binary arithmetic coding (CABAC). The increased degrees of freedom come about by allowing multiple reference frames for prediction and many more tessellations of a 16×16 pixel macroblock.
Video telephony streams are encoded at a lower resolution than entertainment video, but such streams still consume significant bandwidth. Entertainment video often uses the Common Intermediate Format (CIF) with resolution of 352×288 pixels. Video telephony typically employs quarter CIF (QCIF) resolution of 176×144 pixels, for example. QCIF requires approximately 300 kbps for 30 frames per second. If video telephony becomes popular, with many simultaneous users, then 300 kbps would be a high amount of bandwidth for each user. Therefore, a need exists for a cost-effective method and apparatus for bit-rate reduction in video telephony systems.